Displaying 20 results from an estimated 2000 matches similar to: "iax2 qualify - false "peer unreachable""
2006 Nov 01
1
IAX problem
Hi All,
I'm having problem with IAX, I'm trying to connect to speex.co.il from
asterisk using:
register => username:password@speex.dyndns.org
and I cant get it to work.
Maybe someone who already got this to work will help...
When dialing my speex extension I see the next output from consol:
IAX2 Debugging Enabled
*CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:
2007 Apr 03
0
DTMF via IAX ignored after a few seconds
I'm new to this list, and I apologize if this is an already answered
question, but my Google-fu was not strong enough to find the answer if
it was.
I'm having a problem with DTMF on incoming IAX calls. For the first few
seconds of the call (between maybe 1 and 15, it varies from call to
call) everything works fine. After that I continue get DTMF_E messages
from the remote IAX server
2007 Mar 02
1
Double DTMF digits sent on IAX native bridge
Hi,
I have two asterisk servers one is connected to the PSTN and the other
one is connected to SIP users. The two servers connect with each other
using IAX. When I have an incoming call from PSTN to the asterisk
servers and have a forward to go back out to the PSTN the two IAX
channel bridge together. Now every time I dial a DTMF digit, the
asterisk is sending two DTMF digits. I enable
2005 Mar 05
0
Is anybody having problems with sixtel?
Hi,
My termination with sixtel stopped working, is it something I did or anybody
else is having the same problem.
I am attaching log:
*CLI>
-- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack
-- Goto (macro-dialout-default,s,4)
-- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack
-- Goto (macro-dialout-default,s,6)
-- Executing
2007 Sep 14
2
AGI script fails on IAX channels (from call file).
Hi Guys,
I have already tried this one on the developers list. I have not been
successful getting much back there and I have notified them that I will
post this on the users list instead. Hopefully somebody have tried
something similar and can help out.
I am developing AGI scripts on Asterisk and have run into some very
strange behaviour and I think this is a bug, but I am not completely
sure.
2005 Feb 21
2
Conecting to asterisk server through NAT usingIAX
Hallo
Did you allow udp outgoing on 4569 as well.. i found
udp bit different than
tcp when comming to firewalls
liaan
----- Original Message -----
From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws>
To: <timebandit001@gmail.com>; "Asterisk Users Mailing
List - Non-Commercial
Discussion" <asterisk-users@lists.digium.com>
Sent: Monday, February 21, 2005 12:29
2005 Sep 09
0
Transferred calls dropping out of MeetMe
I'm taking inbound calls on an * server, then transferring them to a
second * server where they join a MeetMe conference. If I have
'notransfer=yes' set on the first * server it works fine, but if I
allow the transfer (call then shifts to be between the DID provider
and the second server), the call is dropped 3-5 minutes later.
There is no firewall on my end, and the two
2009 Jun 30
2
IAX2 help needed...
I run a phone in a remote office using the IAX2 protocol. It mostly works
fine; except that every 5 mins it loses connection with Asterisk, before
reconnecting 30 seconds later; rinse & repeat.
Using the IAX2 debugging, I'm seeing this a lot:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE
Timestamp: 00018ms SCall: 04050 DCall: 00000
2005 May 26
1
How do I diagnose the problem in this Asterisk test session with FWD?
=============
SJphone Log
============
Outgoing SIP session
Respondent: (sip:8612@192.168.2.2)
Remote client:
Started: May 26 16:33
Accepted: no
Ended: May 26 16:34
End reason: Call rejected: 503 Service Unavailable
===============
Asterisk Debug
================
Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r")
in new stack
--
2010 Feb 08
0
Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend
of looking for answers.
I have an iax account with Tesco that works flawlessly with the Zoiper
client - but is giving me trouble with inbound calls in Asterisk 1.6.
After some playing I have ended up with an iax.conf file that looks like
this:
[general]
calltokenoptional = 77.75.0.0/255.255.248.0
maxcallnumbers = 16382
2004 Oct 06
0
iax2, strange native bridge problem????
hallo,
i am really confused how nativ briging is working with asterisk,
i use a asterisk server as central server and register another asterisk and
an iaxcomm client to the server, all three have public ips on the internet.
somtimes, when i call from iaxcomm to my asterisk, the calls go peer to
peer (i can see it with tcpdump) but sometimes the get routed through the
central asterisk server
2006 Jun 22
2
iax2 registration problems
On the asterisk1 I got this:
register => username:secret@ipaddress2
[eop]
username=username
secret=secret
type=peer
host=ipaddress1
auth=md5
on the second box I got this
this host is ipaddress2
[incommingiax2]
username=username
type=user
secret=secret
host=dynamic
context=from-internal-custom
auth=md5
on first host 1 am getting:
Jun 22 14:42:10 NOTICE[2398]: chan_iax2.c:7411
2004 Dec 14
0
Codec "Uknown" with IAX connection
I am having some problems getting TelIax service to work with *. Outbound
calls work just fine. When I try an inbound call the phone rings and there
is no audio. Upon further investigation "iax2 show channels" indicates
that the codec is "unknown" The provider confirmed that they are set for
ulaw and so am I. Does anyone have an idea what could be causing the codecs
to
2004 Apr 01
0
I'm still a little lost...
I downloaded iaxComm and get up my iax.conf file and the
extensions.conf. Here is the out but from CLI in iax debug. What did I
forget to do???
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00001ms SCall: 10489 DCall: 00000 [192.168.50.66:4569]
USERNAME : 100
REFRESH : 300
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
2005 May 30
0
IAX2 to H323
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected.
When I dial from SJPhone (H323) ->
2004 Jun 26
1
IAX & FWD, No authority found?
Hi Folks,
Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded:
$IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036
I can make outgoing calls just fine, but when I receive an inbound call
2004 Jul 01
2
IAX2 to IAX2 connection problems
Hi
My head hurts... Can anyone help out here, my remote IAX can see my
local IAX and visa versa, conversation starts, I can dial my remote
(POTS) landline number, remote end answers, trys to route to local
iax2, I see it start the conversation here, the extension (SIP) rings
once and then it dies...
Both ends are defined with accept IPADDRESS to keep it in the family and
simple..
Debug info
2008 Jul 07
1
DTMF on iax channel is not interpreted by asterisk
Hi!
I'm using asterisk 1.4.17 with twinkle and a custom phone based on
iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf.
While the twinkle client is able to initiate an attended transfer using
*2 (as configured in features.conf), the iax client is not. I can see
the DTMF messages showing up on the asterisk console, but asterisk does
not invoke the features
2004 Dec 21
0
IAX2 insists on not using port 4569??
For some reason, starting just today, 1 out 3 of my asterisk servers is
having issues calling 1 other server. The only issue I see is that when
it registers with the problem server it is using port 1027, not 4569.
ie:
Registered to 'Server 1', who sees us as 'Server 2':1027
Server 1 then proceeds to timeout trying to register with Server 2.
The way I have each server registering
2009 Oct 02
1
IAX2 Call rejected, CallToken Support required
Hi All,
I am using Asterisk 1.4.26.2 and I am getting the following problem
making connections to this server. My other servers are Version 1.2.x
which have no problems and this 1.4.26.2 server can call the other 1.2.x
servers.
The error is:
chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support
required. If unexpected, resolve by placing address 192.168.25.250 in
the